When phones can reach the CUCM server but fail to register, a primary step is to verify the TFTP service is running. Phones obtain their configuration files from the TFTP service. If this service is down, phones cannot download their configuration (including the list of active CUCM servers) and will fail to register, even with correct network connectivity and DHCP options.

To meet the requirements for standards-based, secure encryption for both signaling and media, the correct combination is SIP over TLS (which encrypts the SIP signaling messages) and Secure Real-time Transport Protocol (SRTP) for encrypting the actual voice/video media streams. This is the industry-standard approach for securing VoIP communications.

One-way audio in a CUBE-to-ITSP scenario often occurs when Network Address Translation (NAT) is involved. If the CUBE is behind a NAT device, the media (RTP) streams embedded in the SIP messages may contain private IP addresses that the ITSP cannot reach. Configuring ‘media flow-around’ on the relevant dial-peers allows the media to bypass the CUBE and flow directly between the endpoint and the ITSP, resolving the NAT traversal issue and fixing one-way audio.

For basic Extension Mobility functionality, the ‘Cisco Extension Mobility’ service must be activated. This service manages the login/logout process and device profile association. The ‘Cisco CTIManager’ service is also required because Extension Mobility relies on CTI (Computer Telephony Integration) to program the phone’s settings when a user logs in or out.

A Device Pool is a template-like construct in CUCM. Its primary purpose is to group devices (phones, gateways) and assign them a common set of configurations, such as a Region (for codec selection), Date/Time Group, SRST reference, Calling Search Space for auto-registration, and Media Resource Group List. This simplifies administration by applying settings en masse rather than per device.

UDP port 5004 is the registered, well-known port for the Real-time Transport Protocol (RTP). RTP is the standard protocol for delivering audio and video over IP networks. While RTP can use dynamic ports, 5004 is its default. RTCP typically uses the next odd port (e.g., 5005). SIP uses TCP/UDP 5060 or 5061, and H.245 is an H.323 control protocol.

A Call Handler in Cisco Unity Connection is the fundamental building block for call routing. A standard ‘System Call Handler’ can be easily configured as an auto-attendant: you record greetings and define single-digit touch-tone keys (0-9, *, #) to perform actions like transferring to another extension, a hunt pilot, or another call handler. This makes it the most direct tool for creating a basic menu system.

The Cisco QoS Baseline recommendation and best practice is to mark call signaling traffic with DSCP CS3 (Decimal 24, Binary 011000). Signaling is critical but not as delay-sensitive as voice media. CS3 provides a dedicated class selector for signaling, ensuring it gets prioritized treatment above best-effort (DF) and scavenger traffic but below the Expedited Forwarding (EF) queue used for voice media.

The ‘max-ephones’ command under the ‘call-manager-fallback’ configuration mode on a Cisco IOS router sets the maximum number of IP phones that the SRST gateway can support when the connection to the primary CUCM servers is lost. This is a critical license and resource planning parameter. ‘max-dn’ sets the maximum directory numbers, which is typically higher than the number of phones.

When phones can reach the CUCM server but fail to register, a primary step is to verify the TFTP service is running. Phones obtain their configuration files from the TFTP service. If this service is down, phones cannot download their configuration (including the list of active CUCM servers) and will fail to register, even with correct network connectivity and DHCP options.0